§1.1 Field of the Invention
The present invention concerns audio and video conferencing systems and services. In particular, the present invention concerns determining user perceived delays in audio and video conferencing systems and services.
§1.2 Background Information
The Internet has changed the way people communicate, supporting technologies including emails, text-messages, blogs, tweets, Voice-over-IP (“VoIP”) calls, etc. We are now experiencing the next big change—video telephony. Due to its stringent bandwidth and delay requirements, for years, business customers have been paying high prices to utilize specialized hardware and software for video encoding, mixing and decoding, and dedicated network pipes for video distribution. Video telephony has had little success in the end-consumer market, until very recently. The proliferation of video-capable consumer electronic devices and the penetration of increasingly faster residential network accesses has paved the way for the wide adoption of video telephony. Two-party video chat and multi-party video conferencing services are now being offered for free or at low prices to end-consumers on various platforms. Notably, Apple iChat, Google+ Hangout, and Skype Video Calls are among the most popular video conferencing services on the Internet.
Video conferencing requires high-bandwidth and low-delay voice and video transmission. While Skype encodes high quality voice at data rate of 40 kbps, a Skype video call with good quality can easily use up bandwidth of 900 kbps. Although seconds of buffering delay is often tolerable even in live video streaming, in video conferencing, some have argued that user Quality-of-Experience (“QoE”) degrades significantly if the one-way, end-to-end, video delay exceeds 350 milli-seconds. To deliver good conferencing experience to end-consumers over the best-effort Internet, video conferencing solutions have to cope with user device and network access heterogeneities, dynamic bandwidth variations, and random network impairments, such as packet losses and delays. All these have to be done through video generation and distribution in real-time, which makes the design space extremely tight.
Therefore, it would be useful to be able to measure end-to-end delays in video conferencing services, such as iChat, Google+, and Skype, for example. Such measurements could be used investigate how the various video conferencing services address video conferencing challenges and how well they do it on the Internet. However, it is challenging and ambitious to draw conclusions. Various different video conferencing services use proprietary protocols and encrypt data and signaling messages. There is limited public information about their architectures and protocols. As should be appreciated from the foregoing, it would be useful to be able to determine user perceived delays in audio and video conferencing systems and services.